I have been using WebRTC-trial since it first came out last October, I am using it for a video chat application running on Chrome, using Publish/Play demo code. The code was working well last month.
Recently I have encountered the problem that sometimes audio does not play on side of the chat (although video frames still show up). It started happneindg when I was using Wowza 126.96.36.199, and still happens even after I have upgraded to 4.7.0. I am also using the latest Publish/Play demo code (with support for video framerate), using websocket (wss://) connection.
My test setup uses two MacBook laptops side-by-side as clients, running Chrome 57 and 58.
After further investigation, I discover that sometimes the SDP package that the Play code receives from the server might not contain audio BUNDLE (e.g.: a=group:BUNDLE video). Only when the SDP package contains both video and audio then I can hear the audio from the other side (i.e. a=group:BUNDLE video audio). This happens somewhat randomly, as I can start Publish the video on one side, then start Play on the other side, then sometimes the SDP returned from the server for the Play side contains audio BUNDLE, sometimes it does not, even on the same Publish stream.
Right now the work-around I am using is to keep retrying Play until I receive a SDP with audio bundle, otherwise I will just disconnect and reset the connection. It works, but I guess this should be resolved on the server side, why do I not always receive BUNDLE audio+video?
Example of the returned SDP that does not contain audio bundle:
o=WowzaStreamingEngine-next 881639558 2 IN IP4 127.0.0.1
m=video 9 RTP/SAVPF 97
c=IN IP4 0.0.0.0
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir